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Asked: 2026-05-11T13:48:14+00:00

I want to build a VoIP test environment for a student project. It should

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I want to build a VoIP test environment for a student project. It should use SIP.

My first idea is to use Asterisk, OpenSIPS or FreeSWITCH for the server and SIP Communicator, Minisip or Linphone as softphone clients.

  • Does anybody have experience with setting up such an environment?
  • Which combination should I use?
  • Are there any tutorials for setting up this infrastructure?
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  1. I had some experience with Asterisk and SJPhone.

    If your network does not need complex dialing plans (like hotlines etc), and your PC‘s are accessible from each other, then you don’t need a server at all.

    SJphone can establish a peer-to-peer connection just by using an IP address or a host name.

    If your PC’s cannot access each other, then you should setup Asterisk and enable client registration on it.

    On my Fedora 10, Asterisk worked out of box.

    Client registration is like being online on ICQ or MSN: the server knows you’re online and keeps your TCP stream alive in case someone wants to call you.

    It also can be used to show status of people: who’s online, offline, busy etc.

    There is a sample sip.conf file included, it’s really simple. For each client, you create an entry like this:

    [john] type=friend secret=secretpassword host=dynamic

    There are plugins that can keep this data in the database, but I didn’t look at them yet.

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